The present invention relates to a signal processing method and a signal processing apparatus, and more specifically to a method and apparatus for transforming N-bit coded data such as audio signals, video signals, etc. into M-bit coded data on condition that M is larger than N.
In the case where signals such as audio or video signals are converted into digital signals, digital signals having a predetermined number of bits are generated per sample in conformity with a standard prescribed under due consideration of various conditions (e.g., transmission, fidelity of recording and reproduction, apparatus cost, etc.). For instance, in the case of a compact disk, 16-bit digital signals are recorded per sample.
In FIG. 1, a plurality of thick solid lines S as a.fwdarw.b .fwdarw.c.fwdarw.d.fwdarw. . . . k.fwdarw.l.fwdarw.m.fwdarw.n, represent digital signals obtained by quantizing an original analog signal (by resolution of 1/2.sup.N per specific sampling period Ts) in a form of analog signal. Here, the original analog signal resides in the ranges enclcosed by dashed lines including the solid lines S shown in FIG. 1. In other words, there exists an error of less than .+-.0.5 LSB (the least significant bit) between the original analog signal and the restored analog signal (obtained by restoring the original analog signal). Therefore, in the N-bit digital signal obtained by converting the analog signal in the resolution of 1/2.sup.N, the minute signal as fine as obtained by the resolution more that 1/2.sup.N cannot be restored. Further, in FIG. 1, t1, t2, t3, . . . denote sequential sampling points, and Ts denotes a sampling period.
However, there exists so far a need of restoration of the minute signal by a resolution more than a value determined by the number of bits of digital signals. Therefore, for instance, Japanese Laid-Open Patent No. 5(1993)-304474 has proposed such a method of transforming N-bit coded data into M-bit coded data on condition that M is larger than N. In a technique of increasing the number of bits disclosed in this patent, digital signals are smoothed through a digital low-pass filter so that even the minute level signals can be converted by Digital-to-Analog (DA) conversion without distortion. In other words, data less than one LSB of the original number of bits can be output for DA conversion.
In this technique, although the N-bit digital signals are converted into M-bit digital signals (M&gt;N) by use of the digital low-pass filter, it is impossible to correct the errors of 0.5 LSB involved in the N-bit digital signals. In addition, since the digital low-pass filter is used for waveform smoothing, the signal waveform changes. As a result, when this method is applied to the DA conversion of audio digital signals, for instance, there exists a problem in that the quality of the audio digital signals varies.
In order to solve this problem, the applicant of the present application has proposed the following method:
In the case where N-bit coded data obtained by converting analog signals into digital signals in the resolution of 1/2.sup.N are converted into M-bit coded data, n-bit coded data to be subjected to bit number conversion that sequentially appear by one sampling period are detected with respect to their transitions.
In detail, firstly, it is determined on two continuous first and second N-bit coded data whether a digital value of the first N-bit coded data that precedes the second coded data on the time axis is larger, smaller than or equal to that of the second coded data to generate the first (larger), the second (smaller) and the third (equal) detection outputs, respectively.
Secondly, it is determined on the output train of the sequential first and second detection outputs whether a transitional pattern of every four outputs (one output group) is classified into which of predetermined transitional patterns of digital value.
Thirdly, the following operation is executed on N-bit coded data of a plurality of the sequential output groups, each group being composed of the sequential first to fourth digital value transition points, to conduct linear interpolation by means of the digital signal in the resolution of 1/2.sup.N : a linear interpolation to be applied in a period between the second and the third digital value transition points in one output group is decided with respect to the linear interpolation already applied in a period between the first and the second digital value transition points.
Then, an additional signal of (M-N) bits is obtained by means of the digital signal in the resolution of 1/2.sup.N in the following way: the linear interpolation is applied in the period as described above such that an area of a rectangle formed per digital transition point on the time axis in which digital value transition corresponds to one least significant bit (LSB) of the resolution of 1/2.sup.N is almost equal to an area of a figure formed by the rectangle and a line of the digital signal in the resolution of 1/2.sup.N. The additional signal of (M-N) bits is added to an N-bit coded data at its LSB to generate an M-bit coded data.
The above method is illustrated by FIG. 2A. In detail, 1 LSB of the resolution of 1/2.sup.N is composed of points w.fwdarw.x.fwdarw.a.fwdarw.u.fwdarw.v.fwdarw.r.fwdarw.y.fwdarw.z. The linear interpolation is executed on the original digital signal so that a triangle of the points a.fwdarw.u.fwdarw.i and a triangle of the points r.fwdarw.v.fwdarw.i become equal to each other to obtain the digital signal in the resolution of 1/2.sup.N. The additional signal of (M-N) bits is obtained by means of the digital signal in the resolution of 1/2.sup.N. Therefore, an information signal can be obtained that is highly quality compared to a digital signal formed by a conventional bit conversion technique.
The triangles of the points a.fwdarw.u.fwdarw.i and the points r.fwdarw.v.fwdarw.i are equal to each other as described above. However, an actual digital signal in the resolution of 1/2.sup.N is not obtained so as to correspond to the line a.fwdarw.r shown in FIG. 2B but discretely obtained on the time axis per sampling period. Therefore, the triangles of the points a.fwdarw.u.fwdarw.i and r.fwdarw.v.fwdarw.i are actually become polygons b.fwdarw.c.fwdarw.d.fwdarw.e.fwdarw.f.fwdarw.g.fwdarw.h.fwdarw.u and r.fwdarw.v.fwdarw.i.fwdarw.j.fwdarw.k.fwdarw.m.fwdarw.n.fwdarw.p.fwdarw.q, respectively, as shown in FIGS. 2A to 2C. FIGS. 2B and 2C are enlarged views of portions shown in FIG. 2A.
There is room for improvement in this method because the two polygons have different areas therebetween. Therefore, the applicant of the present application has proposed a information signal processing apparatus to modify the additional signal of (M-N) bits with an offset value that makes shift, by one half of the sampling period, the step-like waveform of the digital signal in the resolution of 1/2.sup.N is obtained by linear interpolation.
However, in this method, linear interpolation cannot be executed when N-bit coded data to be subjected to bit conversion sequentially appear per sampling period Ts have various digital values per sampling period.
Therefore, There is still room for improvement in this method.